Tuesday, August 15, 2017

I've
noticed that, when I watch old concert footage, the singer is often
using two mics. I'd always assumed that one was being used for recording
purposes and one was being fed to the PA. However, I recently heard
that the Grateful Dead used a two‑mic technique for noise cancelling, is
this true, and how would it work?

Using
two mics for noise cancellation. Combining outputs of both at equal
gains, but in opposite polarities, will cancel ambient noise from each
to a certain degree.

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: You are quite right in
that, back in the 1970s, it was quite common when recording a live
concert for the recording people to simply tape a second mic to the PA
vocal mic to acquire their recording feed. Mic splitters either weren't
trusted back then or they were too expensive!

The two‑mic noise‑cancelling idea is well known and very common in
many sound applications — most notably aircraft communication headsets —
but is rarely seen these days in live sound and PA. The same basic
physics explains why football commentators' lip ribbon mics work so well
at rejecting the crowd noise.

The basic idea of the two‑mic technique is to have two microphones
spaced a short distance apart (usually between one and two‑inches or two
to six centimetres) in front of the mouth, or whatever the sound source
is. Both microphones must be able to hear the sound source directly. If
they are cardioids, they both need to face the sound source, although
this is more usually done with omnidirectional microphones, for the
following reasons. The ambient noise, being inherently diffuse sound,
will be captured equally in level by both mics; their spacing will make
no significant difference to the ambient sound level they capture. By
contrast, the wanted sound will be in the near field of both mics and,
provided the front mic is very close to the sound source (ie. near the
lips of the vocalist), the inverse square law of sound‑energy dispersion
means that the more distant mic will receive significantly less energy
from the close sound source than the front mic will.

Accordions
produce sound from two places — on most models from 'treble' and 'bass'
ends — so it's important to recognise this when miking and set two
microphones either side and slightly forward of the instrument.

This
also helps to explain why omni mics are preferred in this role, because
otherwise the close mic would have a far stronger bass response, due to
the proximity effect, than the more distant mic, and odd tonal effects
could result. By combining the outputs of both mics at equal gains but
in opposite polarities, the similar level of ambient noise from each
will cancel to a very large degree, whereas the significantly different
levels of the wanted close sound from each mic will hardly cancel at
all. Of course, there will be a slight level reduction of wanted sound
in comparison to using just the close mic on its own but, given the
30dB‑plus of ambient noise reduction gained by this technique, that's
usually a side‑effect well worth suffering when working in very noisy
conditions.

The physical spacing between the two mics inherently introduces
a small, but finite, time delay, and so when the two mic signals are
mixed together, the frequency response will inevitably become comb
filtered. However, if the distance between mics is only an inch or so,
the first deep comb‑filter notch will be well above any significant,
important component of the human voice, and the rest won't have any
material effect on the sound quality either. To return to your original
statement, this noise‑cancelling technique really requires two identical
mics spaced a precise distance apart. Most of those old festival
concert photos show completely dissimilar mics mounted with their
capsules more or less coincident, which lends weight to the suggestion
that they were for separate recording and PA feeds, rather than exotic
noise‑cancelling techniques.

The Grateful Dead developed a version of this noise‑cancelling
technique because of the very unconventional PA arrangements they used
to employ, with all of the PA set up on stage behind the band as a 'wall
of sound'. In this way the band heard exactly what the audience heard
(no need for separate monitors!). It was a clever system, with each
musical source having its own set of amps and speakers to improve
headroom and minimise distortion.

Saturday, August 12, 2017

I'm trying to learn a little more about amp design. One thing that
really baffles me is the different classes available. What does an amp's
class mean, and how does this affect the way it is used?

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: In a Class‑A circuit, the
active device (whether valve or solid‑state) passes current regardless
of the polarity of the input signal; in other words, in an audio
application, it is 'biased' so as to pass both the positive cycle and
the negative cycles of an audio signal. The side effect of the biasing
is that the active device has to pass current all the time, making it
relatively inefficient.

In a Class-B circuit, the active device only passes current for one
polarity of input signal — which polarity depends on the circuit design —
and this makes it a much more efficient way of working. So, in this
case, where it is required to pass a symmetrical audio signal using
a Class‑B circuit, the circuit will need two active devices, one to
handle each polarity. This is an arrangement often also known
as 'push‑pull'.

Class C is a format that only conducts on signal peaks and is rarely
(but occasionally) used for audio in situations where power efficiency
is more important than distortion. Class D — which is now becoming very
popular in audio applications — works by generating a stream of
high-voltage pulses at a very high frequency. These pulses are modulated
in such a way that the average energy they convey follows the wanted
audio waveform.

Returning to the Class-B design, this exhibits a problem called
crossover distortion for audio applications, because both of the active
devices in the push‑pull pair turn off as the signal nears the zero
line. The solution is to bias the devices so that they don't turn off.
They actually continue to pass signal as it crosses over into the
opposite polarity. In other words, it works a little more like
a Class-A device (but without the same levels of power inefficiency).

In a push‑pull amp design, each active device handles one polarity of the input signal.

Hence the compromise name Class AB; it is a Class-B design biased to
operate in a similar way to Class A around the crossover region.
However, it should also be remembered that push‑pull designs can also be
operated fully as Class A if required, and some high‑power amps do work
in that way. This is also a handy technique for cancelling out
even-harmonic distortion products in tube-amp designs.

Thursday, August 10, 2017

The diagram to the right shows my room, which serves as my studio.
The dimensions seem to be bad for low frequencies and there are
sound-pressure failures at 55Hz and between 110 and 140 Hz. I have an
Auralex foam bass trap, but I don't known if absorption is the answer.
What should I do to improve this situation?

If you have any choice of rooms for your studio, try to avoid those whose dimensions are multiples of each other.

Via SOS web site

SOS columnist Martin Walker replies: I agree: that's a bad choice for
a room, dimensionally, as far as acoustics are concerned. The 2.6‑metre
width and 2.5‑metre height are nearly identical, while the 5.8‑metre
length is close to double these, giving you a shape that's almost two
cubes joined together. The room is also relatively small, which will
mean it'll have relatively few modes below a few hundred hertz and, as
the dimensions are closely related to each other, these modes will pile
up at some frequencies (resulting in a huge peak), with large gaps
between them (creating big dips in the frequency response).

Room-mode frequencies are fairly easy to calculate, but it's even
easier to plug your three dimensions into a utility, such as the on‑line
MCSquared Room Mode Calculator (www.mcsquared.com/metricmodes.htm) or the Hunecke Room Eigenmodes Calculator (www.hunecke.de/en/calculators/room‑eigenmodes.html). However, if you've got a PC, the ModeCalc utility from Realtraps (www.realtraps.com/modecalc.htm)
is one of the easiest to use, displaying the first 16 axial modes for
each room dimension up to 500Hz in an easy-to-interpret graphics plot.
It would show that the biggest gaps in your room mode plot occur between
30 and 60 Hz (which explains your hole at 55Hz), between 70 and 90 Hz,
and again between 90 and 130 Hz (the other area you've
already pinpointed).

Without acoustic treatment, your listening position will be very
critical, since you can end up sitting in a bass trough at one frequency
and a huge peak at another. However, your loudspeakers and listening
position do look to be near their optimum locations for the flattest
compromise response. The oft‑quoted ideal is to place your listening
position (ears!) close to 38 percent into the room from the front wall.

Acoustic-foam bass traps, like the one you already have, can
certainly be effective, and acoustic foam is also excellent for dealing
with mid-/high-frequency early reflections from your side walls and
ceiling. However, acoustic foam is invariably a lot less dense than the
60kg/m3 Rockwool that is generally recommended for DIY bass traps, and
you will, therefore, require a much greater volume of it to achieve
a similar amount of absorption at lower frequencies. In a small room,
you'll simply run out of space before you can cram in enough acoustic
foam traps to adequately deal with the problems.

Diffusion can be a good way to 'break up' your reflections so they
become less troublesome, but you ideally need to be at least a couple of
metres away from them to avoid hearing a set of discrete reflections,
rather than a more diffuse soundfield, so they are not often used in
small rooms like yours. Tuned traps also have their place in the grand
scheme of things but, in my experience, tend to be more difficult to
tune and place optimally compared with broadband trapping that you
simply fit where the bass levels are loudest, so they absorb the sound
more efficiently.

Overall, I think broadband absorption is your best bet; as much of it
as you can reasonably fit into your room. Start by placing traps that
straddle the front vertical corners of the room, then the rear vertical
corners, followed by any other corners you can manage, such as the
ceiling/wall corners, and even the floor/wall corners where feasible.
Also, don't forget some side panels and ceiling 'cloud' at the 'mirror
points' to deal with early reflections.

Friday, August 4, 2017

I've
been reading about how you have to be quite precise in matching the
distance from source to mic when multi‑miking guitar cabinets, and
something occurred to me. If this kind of phase alignment is so
important in this instance, how can we avoid such issues when
double‑tracking a vocal, given that the singer inevitably moves their
head around? The singer in question here is me, and I tend to move
around a fair bit when singing! I've noticed when lining up and trimming
my doubled vocals in the past (and on my current song) that some words
sound 'different' when combined than others, and by different I mean
'worse'. Could phasing be the underlying cause, and if so, is there
anything I can do to rectify this?

Sorting
your sample library into nested folders is an excellent way to help you
find what you're looking for more quickly, but some software samplers
(like NI's Kontakt 4, shown here) already provide extensive database
'tagging' systems for just that purpose.

Via SOS web site

SOS contributor Mike Senior replies: Yes, if you double‑track very
closely, you'll inevitably get some phase‑cancellation between the two
layers, but that's not a problem; it's an inherent part of what makes
double‑tracking sound the way it does. However, the potential for phase
cancellation between the parts won't be nearly on the same scale as with
the two signals of a multi‑miked guitar amp, because, firstly, the
waveforms of two different vocal performances will never match anywhere
near as closely; and, secondly, the phase relationship between the
performances will change from moment to moment, especially if you're
moving around while singing. Furthermore, in practice a vocal
double‑track often works best when it's lower in level than the lead, in
which case any phase‑cancellation artifacts will be much less
pronounced.

For these reasons, nasty tonal changes from double‑tracking haven't
ever really presented a major problem for me, and if they're regularly
causing you problems, I suspect you might be trying to match the layers
too closely at the editing stage. Try leaving a little more leeway for
the timing and see if that helps for a start — just make sure that the
double‑track doesn't aniticipate the lead if you don't want it to draw
undue attention to itself. Similarly, try to keep pitch‑correction as
minimal as you can (especially anything that flattens out the
shorter‑term pitch variations), because that will also tend to match the
exact frequency of the two different waveforms. In fact, if there are
any notes that sound really phasey to you, you might even consider
shifting one of the voices a few cents out of tune to see if that helps.
Anything you can do to make the double‑track sound less similar to the
lead can also help, whether that means using a different singer (think
Lennon and McCartney), a different mic, or a different EQ setting. You
may only need the high frequencies to provide the double‑tracking
effect, and these are unlikely to phase as badly as the low frequencies.

Wednesday, August 2, 2017

I've
carefully wired up my gear using all balanced inputs and outputs, and
proper balanced cables, but I'm still getting occasional digital hash in
the background. What have I missed?

Even
with balanced cables you can sometimes experience ground loops, so
here's the best place to break one without risking RF interference.

Jamie, via email

SOS columnist Martin Walker replies: Ground‑loop problems can be
absolutely infuriating, and I wrote a step‑by‑step guide to tracking
them down back in SOS July 2005 (/sos/jul05/articles/qa0705_1.htm).
In essence, you have to temporarily unplug all the cables between your
power amp and mixer. If the noises go away, you've found the location of
your problem. If not, plug them back in and try unplugging whatever
gear is plugged into the mixer — and so on down the chain.

The majority of ground‑loop problems occur with unbalanced
connections, so my next advice would have been to replace the offending
unbalanced cable with a balanced or pseudo‑balanced version. However, as
you've found, sometimes such problems occur even in fully balanced
setups where you carefully connect balanced outputs of one device to
balanced inputs of another via 'two‑core plus screen' balanced cables.
I recently had just such a problem in my own studio and, to make it
even worse, it was an intermittent one, so whenever I got close to
discovering its cause, it mysteriously vanished again. Here's what I did
to track it down, so others can try some similar detective work in
their own setups.

First of all, you've got to be systematic, and note down everything
you try, particularly with an intermittent problem, so you don't have to
start from scratch every time it occurs. In my case, I could hear the
digital low‑level hash through my loudspeakers even with my power‑amp
level controls turned fully down, and it also persisted when I turned
off the D‑A converter box feeding my power amp. However, it completely
disappeared as soon as I disconnected both cables between the D‑A output
and power amp input.

These quick tests confirmed that the noise wasn't coming from the
output of the converter, or from the power amp itself, but instead from
a ground loop completed when the two were connected. However, just like
you, I was already using balanced cables. I double‑checked the wiring of
both of my XLR balanced cables and there were no errors: the screen of
the cable was connected to pin 1 at each end, the red core connected to
pin 2 at each end, and the blue (or black) core to pin 3 at each end. So
far, so good.

Next, I double‑checked with a multimeter that there was no electrical
connection between the metalwork of the two devices via my equipment
rack (a common source of ground‑loop problems, and curable by bolting
one of the devices to the rack using insulated washers or 'Humfrees').
Again, there was no problem.
The best wiring for balanced audio equipment is to tie the cable
screen to the metal chassis (right where it enters the chassis) at both
ends of the cable, which guarantees the best possible protection
from RFI (Radio Frequency Interference). However, this assumes that the
interconnected equipment is internally grounded properly, and this is
where things can go awry. The cure is to disconnect one end of the cable
screen, and the best choice to minimise the possibility of RFI is the
input end (as shown in the diagram).

By this time, my intermittent problem had disappeared again, so
here's another tip. I carefully cut the screen wire of one of my two
cables just before it arrived at pin 1 of the XLR plug, but left the
other cable unmodified. Then, the next time the ground loop problem
occurred a few days later I quickly unplugged the unmodified cable,
whereupon the noise disappeared immediately. This proved that I'd
correctly tracked down the problem, and modifying the other cable in the
same way ensured that it never happened again.

Tuesday, August 1, 2017

What
differences can you hear when comparing inexpensive and expensive
equipment? As I do a lot of vocal recording, I'd like to splash out on
a really good microphone. But how can I be sure that an expensive
microphone is worth the money? What am I listening for?

Sarah Betts, via email

Fidelity
and accuracy are expensive qualities to build into a microphone, so
those are the areas that will generally improve as you increase your
budget. However, this doesn't necessarily mean that your voice will
sound better through a more expensive mic; it's more important that you
find the right mic to suit you. Bono, for example, famously favours the
inexpensive Shure SM58 over high‑end alternatives.

SOS Technical Editor Hugh Robjohns replies: The
benefits extend far wider than just the sound, but basically you're
listening for an improvement over your current mic, and you then need to
decide if the price justifies that improvement, bearing in mind the law
of diminishing returns. Going from a very low‑budget mic to a mid‑range mic will usually bring about very obvious sound improvements. Going from there to a high-end model will bring smaller improvements, which may not always be obvious. And going from there to a mic worth several thousand dollars will bring smaller benefits still. Some people will believe the improvements are worth the expense, others won't!

However, you'll know immediately and quite instinctively when you
find a mic that is well suited to your voice, and that doesn't always
mean the mic needs to be expensive. If you're looking for
a general-purpose mic, expensive usually equates to increased
flexibility in use. But if it's a mic that will always be used on your
voice and nothing else, finding a mic that suits your voice is the prime
directive.

Sonic fidelity or accuracy is generally an expensive thing to
engineer into a microphone, and the most expensive mics are generally
pretty accurate. But recording vocals is rarely about accuracy. It's
more to do with flattery, and different voices need to be flattered in
different ways. When working with a new vocalist, I'll usually try
a range of mics to see which one works best with their voice. Sometimes
the most expensive mic gives the best results, but it's equally likely
that it will be a less expensive model. U2's Bono famously records his
vocals using a Shure SM58, and he seems happy with the results!

But, as I said, there's more to an expensive mic that just the sound.
More expensive mics tend to be built to higher standards. They tend to
include internal shock-mounting for the capsule, to reduce handling
noise. They are thoroughly tested to comply with the design
specifications and provide consistent results. Being better constructed,
they tend to have longer working lives and can be maintained by the
manufacturer relatively easily. They also generally deliver a very
usable (although that might not necessarily equate to 'the best') sound
whatever the source, without needing much EQ to cut through in the mix.

Less expensive mics often sound great on some things but terrible on
others, often needing a lot of EQ to extract a reasonable sound within
a mix. Often they're less well manufactured, which reduces their working
life expectancy and, once broken, can rarely be repaired.

Monday, July 31, 2017

I'm
a synth guy getting more and more into recording and mixing my own
tunes. One thing that stumps me is the issue of 'headroom': for example,
in the case of my Focusrite Saffire Pro 26 I/O, the manual says that
using the PSU rather than Firewire bus power yields 6dB of additional
headroom in the preamps. I assume that this is a good thing, but how so?
What is headroom and why do I want more of it? How do I know it's there
(or not there), and how can I take advantage of it?

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: These are all good
questions. Every audio‑passing system (analogue or digital) has two
limits: at the quiet end there is the noise floor, normally
a constant background hiss into which signals can be faded until they
become inaudible; and at the loud end there is clipping, the point where
the system can no longer accommodate an increase in signal level and
gross distortion results. The latter is generally due to the signal
level approaching the power supply voltage levels in analogue systems,
or the coding format running out of numbers to count more quantising
levels in digital systems.
Obviously, we need to keep the signal level somewhere between these
two extremes to maximise quality: somewhere well above the noise floor
but comfortably below the clipping point. In analogue systems, this is
made practical and simple by defining a nominal working level and
encouraging people to stick to that by scaling the meters in a suitable
way. For example, VU meters are scaled so that 0VU usually equates to
+4dBu. The clipping point in professional analogue gear is typically
around +24dBu, so around 20dB higher than the nominal level indicated on
the VU meter.

That 20dB of available (but ideally unused) dynamic‑range space is
called the headroom, or is referred to as the headroom margin. It
provides a buffer zone to accommodate unexpected transients or loud
sounds without risking clipping. It's worth noting that no analogue
metering system displays much of the headroom margin. Rather, it's an
'unseen' safety region that is easy to overlook and take for granted. In
most digital systems, the metering tends to show the entire headroom
margin, because the meter is scaled downards from the clipping point at
0dBFS. The top 20dB or so of a digital scale is showing the headroom
margin that is typically invisible on the meters of analogue systems. As
a result, many people feel they are 'under‑recording' on digital
systems if they don't peak their signals well up the scale, when in fact
they are actually over‑recording and at far greater risk of transient
distortion.

The reason why your interface offers greater headroom when operating
from its external power supply is because the PSU provides
a higher‑voltage power rail than is possible when the unit is running
from the USB power supply. A higher supply voltage means that a large
signal voltage can be accommodated; in this case, twice as large, hence
the 6dB greater headroom margin. More headroom means you have to worry
less about transient peaks causing clipping distortion, and generally
translates to a more open and natural sound, so it's a good thing.

Saturday, July 29, 2017

I'm
trying to figure out how I would create a really old‑style,
warm‑sounding distortion/crackle on a string motif for an intro to
a song I'm writing. I'll be using East West Quantum Leap Symphonic
Orchestra for the actual string loop, and I want to create a sort of 'AM
radio' feel for it. That's easy enough to achieve using various EQ
techniques, but I also want to give it a really subtle '60s
record‑player crackle — something that's there if you know what you're
listening for, but not so 'in your face' as to sound cheesy or clichéd.
I was wondering if there are plug‑ins that can do this. I fear I may
have to break the bank again...Here
are three plug‑ins you could use to add simulated vinyl noise to your
audio tracks without breaking the bank: Izotope's Vinyl (left), Retro
Sampling's Vinyl Dreams (far left), and Steinberg Cubase's bundled
Grungelizer (top).

Via SOS web site
SOS contributor Mike Senior replies: There's no need to break the
bank for this, because there are actually a few different freeware
plug‑ins that provide the kind of thing you're after. One of the best
known is Izotope's freeware Vinyl plug‑in, which is available for both
Mac and PC. The advantage of this one is that you get a lot of control
over the exact character of the vinyl noise you're creating: not only
can you balance various different mechanical and electrical noises, but
you can also choose the decade you want your virtual vinyl to hail from
and how your processed audio is affected by disc wear.

The downside of
this plug‑in for me, though, is that it doesn't seem to output some of
its added noises in stereo, irrespective of how I set up the controls,
and a lot of the character of vinyl noise, to me, lies in its stereo
width. To be fair, though, the 'dust' and 'crackle' components seem to
be stereo, and stereo was, of course, only really in its infancy in the
'60s, so this might not matter to you. Indeed, collapsing the whole
signal to mono might be a useful way to 'date' the string sound itself.
If you're running Steinberg's Cubase, the built‑in Grungelizer plug‑in
provides a similar paradigm to the Izotope plug‑in, albeit with
a simpler control set. However, all the added noises from this plug‑in
appear to be in mono too.
For stereo vinyl noise, check out the freeware plug‑ins from Retro Sampling (www.retrosampling.se).
Both Audio Impurities and Vinyl Dreams can overlay vinyl noise,
although you only get wet/dry knobs, so you're stuck with the preset
effect. That said, if you set up the plug‑ins on a separate channel in
your sequencer, you can dramatically adjust their character with EQ to
make them seem less obtrusive — a combination of high‑cut and low‑cut
filtering usually works well for me. If you want a smoother vinyl noise
(less of the Rice Crispies!), you can also slot in a fast limiter or
dedicated transient processor to steamroller spikes in the waveform.

These processing techniques also allow you to get good mileage from
the vinyl noise samples that periodically crop up on sample libraries.
I've been collecting vinyl noise samples for a while, so I can tell you
that there are good selections on the Tekniks Ghetto Grooves and Mixtape
Toolkit titles, as well as on Spectrasonics' original Retrofunk
collection. I've also turned up a good few examples in general‑purpose
media sound‑effects libraries, if you have anything like that to hand.

Thursday, July 27, 2017

I record and mix in my 'studio',
which isn't too great acoustically. I can manage somehow when mixing, by
working on headphones and doing lots of cross‑referencing, but the problem is
that when it comes to recording I really hate the room sound on my vocals,
and most of all on acoustic guitars, which I use a lot. The reverb
tail is pretty short, but I'm still having a hard time getting a nice
dry sound on my guitars, because I can't record dry! I know that the
obvious solution is to treat the room, but the truth of the matter is that
I can't do much better than this for now. So is there any way to treat
a 'roomy' sound (on vocals and guitar) to make it sound drier? I know
it is very difficult, or maybe impossible, especially for acoustic guitars, but
any kind of suggestion, even for small improvements, would be very welcome.

A
high‑resolution spectrum analyser such as Schwa's Schope lets you quickly and
precisely home in on specific resonant frequencies that may be responsible for
a coloured or uneven sound.

Via SOS web site

SOS contributor Mike Senior replies:
Given that the reverb doesn't have a 'tail' as such, I reckon it's
the reverb tone that's the biggest problem, so trying to use some kind of
gating or expansion to remove it is unlikely to yield a useful
improvement. You could help minimise the ambient sound pickup by using
a directional mic for both vocals and guitar and keeping a fairly
close placement. For vocals, very close miking is pretty commonplace, but for
acoustic guitar you might want to experiment with using an XY pair of mics
instead of a single cardioid, to avoid 'spotlighting' one small area of
the guitar too much. That setup will usually give you a more balanced
sound because its horizontal pickup is wider than a single cardioid on its
own. In all but the smallest rooms, it's usually possible to get
a respectable dry vocal sound just by hanging a couple of duvets
behind the singer, and because I suspect that you've already tried this
fairly common trick, I'm suspicious that room resonances are actually the
biggest problem, rather than simple early reflections per se. Duvets are quite
effective for mid‑range and high frequencies, but aren't too good at dealing
with the lower‑frequency reflections that give rise to room resonances.

So given that room resonance is
likely to be the problem, what can you do about it? Well, if you've no budget
for acoustic treatment, I'd seriously consider doing your overdubs in
a different room, if there's one available. If you're recording on
a laptop, or have a portable recorder, maybe you can use that to
record on location somewhere if you're confined to just the one room at home.
I used to do this kind of thing a lot when I first started doing
home recordings, carting around a mic, some headphones and a portable
multitrack machine to wherever was available.

Part of what the room resonances
will be doing is putting scary peaks and troughs into the lower mid‑range of
your recorded frequency response, but the exact frequency balance you get will
depend on exactly where your player and microphone are located in relation to
the dimensions of the room, so a bit of determined experimentation in this
respect might yield a more suitable sound, if not quite an uncoloured one.
You might find that actually encouraging a few more high‑frequency early
reflections using a couple of judiciously placed plywood boards might also
improve the recorded room sound a little. A lot of domestic
environments can have a bit too much high‑frequency absorption, on account
of carpets, curtains, and soft furnishings.

After recording, you could also get
busy with some narrow EQ peaks in the 100‑500Hz range, to try to flatten any
obvious frequency anomalies. One thing to listen for in particular is any notes
that seem to boom out more than others: a very narrow notch EQ aimed
precisely at that note's fundamental frequency will probably help even things
out. You can find these frequencies by ear in time‑honoured fashion by sweeping
an EQ boost around, but in my experience a good spectrum analyser like
Schwa's Schope plug‑in will let you achieve a better result in
a fraction of the time. However, while EQ may address some of the
frequency‑domain issues of the room sound, it won't stop resonant frequencies
from sustaining longer, which is just as much part of the problem, and there's
no processing I know of that will deal with that.

For my money, this is the kind of
situation where you can spend ages fannying around with complicated processing
to achieve only a moderate improvement, whereas nine times out of 10
you'll get better results much more quickly by just re‑recording the part.

Wednesday, July 26, 2017

I always hear people talking about
low‑pass filters and high‑pass filters and cutting at this and that frequency,
but where do you get these filters from? I don't think I have one in
my Cakewalk Project 5 software. Are they part of equalizers?

Via SOS web site

This
diagram illustrates both low‑pass (high cut) and high‑pass (low‑cut) filtering.
The shaded areas in the diagram will be attenuated.

SOS Technical Editor Hugh Robjohns
replies: People sometimes use the terms 'EQ' and 'filter' interchangeably, so
it's understandable that you might be confused. We've published several
introductory guides to EQ, most recently in SOS December 2008 (/sos/dec08/articles/eq.htm), so if you're a bit
baffled about the broader subject of EQ, it would be well worth reading this.

Essentially, EQ is used to boost or
attenuate (turn down) a range of frequencies in order to shape
a sound. High‑pass and low‑pass filters are common in professional
equalisers, but less common in budget designs. They are used to define the
highest and lowest frequencies of interest in the signal and they pretty much
do what their names suggest: let audio above a certain frequency pass
(high‑pass filter) or audio below a certain frequency pass (low‑pass
filter). Anything outside those limits is attenuated. They are also called low‑cut
or high‑cut filters, but the function is the same.

Filters are defined by their slope, which
determines the attenuation of signals outside the 'pass' band. Most audio
filters on mixing desks (and DAWs) will have a slope of 12dB or 18dB per
octave, and in synthesizer filters the slope may be as steep as 24dB per
octave. If an 18dB/octave high‑pass filter is set to 80Hz, any audio an octave
below that (at 40Hz) will be attenuated by 18dB, and an octave lower still, at
20Hz, it will be attenuated by 36dB... and so on.

High‑ and low‑pass filters generally
have much steeper slopes than the more normal equaliser bands (which are
typically only 6dB/octave) and are intended for a different purpose. You
can't effectively remove rumble with a bass EQ control, but you can with
a high‑pass filter. But equally, you can't shape the tone of a bass
guitar with a high‑pass filter as easily as you can with a bass EQ
control.

Filters are used for 'corrective'
equalisation, as opposed to creative equalisation. They are used to clean up
a signal, rather than to shape the sound creatively. They only provide
attenuation of unwanted frequencies, and there's no scope to boost any part of
the frequency range. Of the two, the high‑pass filter is probably the most
useful, as it helps to remove unwanted rumbles and other unwanted sub‑sonic
rubbish that microphones tend to capture. Most DAW software includes
a software EQ that you'll be able to use to perform any of these tasks,
and although I'm not personally familiar with Cakewalk Project 5, I notice
that it can host third‑party VST plug‑ins, so there are many freeware plug‑ins
that you could use if your DAW doesn't have them built in.